CS 414 - Spring 2009

CS 414 – Multimedia Systems Design Lecture 36 – Voice-over-IP/Skype

Klara Nahrstedt

Spring 2009

Outline

Voice-over-IP Basic Principles Skype – first VoIP over Peer-to-peer

Infrastructure

CS 414 - Spring 2009

Voice over IP (VoIP) VoIP – transport of voice over IP-based networks Complexity ranges from

Hobbyists using Internet to get free phone calls on peer-to-peer basis to

Full scale PSTN replacement networks VoIP must address

Types of end user terminals - IP phones, PC clients Quality of Service – ensure agreed quality Security risks must be clearly identified Last mile bandwidth – which affects codec, packetization period

and where to use compression to best meet service goals Signaling protocol must support service set required

Next Generation VoIP Network (MSF Example)

MSF VoIP

Access Services Signaling protocol and network service signaling protocol: SIP Use RTP packets for telephony events Transport DTMF tones out of band using the signaling

protocol such as SIP

Quality of Service (Delay, Jitter, Packet loss) Use RSVP, DiffServ, MPLS, even ATM RTP is used for media traffic

Skype

Source: An Analysis of the Skype Peer-to-peer Internet Telephony

Protocol, S. Baset, H. Schulzrinne,

2004

Skype Overview Peer-to-peer (P2P) overlay network for Voice-over-IP

(VoIP) and other application Developed by Niklas Zennstrom and Janus Friis

(founders of KaZaA, file-sharing company) Users see Skype as an Instant Messaging (IM) software Free on-net VoIP service and fee-based off-net

SkypeOut service (allows calling to PSTN and mobile phones)

Runs on Windows, Linux, Pocket PC, …

CS 414 - Spring 2009

Skype Network

Super Nodes: Any node with a public IP address having sufficient CPU, memory and network bandwidth is candidate to become a super node

Ordinary Host: this host needs to connect to super node and must register itself with the Skype login server

CS 414 - Spring 2009

Components of Skype

Ports Skype client (SC) opens TCP and UDP listening port

configured in its connection dialog box Host Cache (HC)

List of super node IP address and port pairs that SC builds and refreshes regularly

SC stores HC in the Windows registry Codecs

Wideband coded allowing frequencies between 50Hz-8KHz (one of the codecs is implemented by Global IP Sound)

Skype Ports on which Skype listens for incoming connections

Skype Host Cache List

Components of Skype

Buddy List Skype stores buddy information in Windows registry Buddy list is digitally signed and encrypted, local to machine and

not on a central server Encryption

Skype uses 256-bit AES encryption Skype uses 1536 to 2048bit RSA to negotiate symmetric AES

keys NAT and Firewall

SC uses variations of the STUN and TURN protocols to determine type of NAT and firewall

STUN and TURN

STUN (Simple Traversal of UDP through NAT) Does not work through symmetric NAT

TURN (Traversal Using Relay NAT) Increase latency and packet loss

Techniques used in Skype

Firewall and NAT traversal Global decentralized user directory Intelligent routing Security Super-simple UI

Login

During login process SC: Authenticates its user name and password with login

server Advertises its presence to other peers and its buddies Determines type of NAT and firewall it is behind Discovers online Skype nodes with public IP

addresses Login server is the only central component in

Skype network

Skype Login Algorithm

Skype Login Process

After installation and first time startup, HC was observed empty

Bootstrap supper nodes: After login for the first time after installation, HC was

initialized with seven (IP,port) pairs

Bootstrap (IP,port) information either Hard coded in SC Encrypted and not directly visible in Skype Windows

registry, or One-time process to contact bootstrap node

Skype Login Process

First time Login Process SC sends UDP packets to some bootstrap SNs SC establishes TCP connection with bootstrap SNs that respond SC perhaps acquires address of login server from SNs SC establishes TCP connection with login server, exchanges

authentication information Subsequent Login Process

Similar to first-time login process SC uses login algorithm to determine at least one available peer

and establishes TCP connection HC was periodically updated with new peers’ (IP,port)

Skype Login Process Comparison of three network setups

Exp:A both Skype users with public IP address Exp B: one Skype user behind port-restricted NAT Exp C: Both Skype users behind port-restricted NAT and UDP-

restricted firewall Message flows for first time login process

Exp A and Exp B are roughly the same; Exp C only exchange info over TCP

User Search

Skype uses Global Index technology to search for a user

Skype claims that search is distributed and is guaranteed to find a user if it exists and has logged in during last 72 hours

Search results are observed to be cached at intermediate nodes

Call Establishment and Teardown

Call signaling is always carried over TCP For user not present in buddy list, call placement

is equal to user search plus call signaling If caller is behind port-restricted NAT and callee

is on public IP, signaling and media flow through an online Skype node which forwards signaling to callee over TCP and routes media over UDP

If both users are behind port-restricted NAT and UDP-restricted firewall, both caller and callee SCs exchange signaling over TCP with another online Skype node, which also forwards media between caller and calllee over TCP

Media Transfer and Codec

Bandwidth usage: 3-16 Kbytes/s Skype allows peers to hold a call. To ensure

UDP binding, SC sends three UDP packets per second to the call peer on average

No silence suppression is supported in Skype The min. and max. audible frequencies Skype

codecs allow to pass through are 50 Hz and 8000 Hz.

Uplink and downlink bandwidth of 2KB/s each is necessary for reasonable call quality

Conferencing

Node A acts as mixer, mixing its own packets with those of node B and sending to C and vice versa

For three party conference, Skype does not do full mesh conferencing

Most powerful machine will be elected as conference host and mixer

Two-way call: 36kb/s Three-way call: 54kb/s

Impact of Skype

Impact on fixed-line operator Skype will introduce SkypIN

Impact on mobile phone operator Skype will be embedded in Wi-Fi/mobile phone WLAN is now limited by

Not many Wi-Fi phone models Wi-Fi phone’s high price Batter life Not enough hot-spots

Impact of Skype

Skype has shown, at least has suggested, the following Signaling, the most unique property of

traditional phone systems, can now be accomplished effortlessly with self-organizing P2P networks

P2P overlay networks can scale up to handle large-scale connection-oriented real-time services such as voice

Conclusion

More than 2 million on-line subscribers per day

More than 2.7 billion minutes served (minutes of free Skype-to-Skype calles)

More than 38 million of software download More than 7 million of registered

subscribers More than 1 million concurrently on-line

subscribers